1. Field of the Invention
The present invention relates to hearing aids. The invention further relates to methods of processing sound signals in hearing aids. The invention still further relates to controlling sound signals and, more particularly, to methods and hearing aid devices that process sound signals, in particular for hearing impaired persons by using a multitude of compressors.
In this application, a hearing aid should be understood as a small, battery-powered, microelectronic device designed to be worn behind or in the human ear by a hearing-impaired user. Prior to use, the hearing aid is adjusted by a hearing aid fitter according to a prescription. The prescription is based on a hearing test, resulting in a so-called audiogram, of the performance of the hearing-impaired user's unaided hearing. The prescription is developed to reach a setting where the hearing aid will alleviate a hearing loss by amplifying sound at frequencies in those parts of the audible frequency range where the user suffers a hearing deficit. A hearing aid comprises one or more microphones, a battery, a microelectronic circuit comprising a signal processor, and an acoustic output transducer. The signal processor is preferably a digital signal processor. The hearing aid is enclosed in a casing suitable for fitting behind or in a human ear.
The microphone in the hearing aid converts sounds from the surroundings into an analog, electrical signal. The digital signal processor in the hearing aid converts the analog electrical signal from the microphone into a digital signal by virtue of an analog-to-digital converter. Subsequent signal processing is carried out in the digital domain. The digital signal is split up into a plurality of frequency bands by a corresponding plurality of digital band-pass filters, each band-pass filter processing a separate frequency band. The plurality of band-pass filters is usually denoted a band-split filter. The signal processing in each frequency band comprises gain calculation and compression, compression being required because a hearing impairment is generally associated with a reduced dynamic range. After processing the signal in the separate frequency bands, the plurality of frequency bands are summed before converting the digital output signal into sound.
Digital hearing aids are thus capable of amplifying a plurality of different frequency bands of the input signal separately and independently and subsequently combining the result to extend over a coherent, audible range of frequencies, suitable for acoustic rendering. Part of the amplification process involves a compression algorithm for controlling the dynamics of each band separately, and the amplification gain and compressor parameters may be controlled separately for each band in order to tailor the sound reproduction to a specific hearing loss.
The compressors present in contemporary hearing aids usually have their settings optimized during the procedure of fitting the hearing aid to a user's hearing loss for the purpose of reproducing speech faithfully and comprehensibly. Other sounds are of course reproduced by the hearing aid as well, but the processing quality of speech signals is paramount. Speech signals in noise are particularly difficult to understand by a hearing impaired person, and the optimization process thus takes this factor into account when the hearing aid is fitted to the user.
In this application the term “compressor system” is referred to as comprising a “signal level estimator” and a “compressor”. The signal level estimator is referred to as a circuitry that supplies an estimated signal level to the compressor for use in the compressor as input. The compressor then calculates a signal gain value to be applied in the signal processing based on said input.
Furthermore the term “compression ratio” is referred to as the inverse of the slope of the input-output curve for the hearing aid. This curve illustrates the output sound pressure level as a function of the input sound pressure level. The term “knee point” is referred to as a point on the input-output curve, where the slope changes.
The compression characteristics of the slow and the fast compressor constitute the corresponding input-output function of the slow and fast compressor.
In this application the speed of the signal level estimator is referred to as “fast” when the estimated signal level responds fast to changes in the signal level estimator input signal and therefore follows the input signal relatively closely and is referred to as “slow” when the estimated signal level responds slowly to changes in the signal level estimator input signal and therefore can not follow the input signal fluctuations and becomes some kind of input signal average.
In this application the “envelope signal” is the signal level estimator input signal. The envelope signal is provided by transforming the acoustic input sound signal into an electric input signal, determining the absolute value of the electric input signal, and finally low pass filtering the absolute value of the electric input signal in order to extract the envelope signal.
In this application “attack time” and “release time” of the signal level estimator is a measure of the speed of the signal level estimator. Therefore the attack and release times of the signal level estimator are short when the speed of the signal level estimator is fast. However, in this application these terms “attack time” and “release time” are expressed by values measured in dB/s in order to make the signal level estimator speeds independent of the clock frequency for the signal level estimators. With this choice of units the speed of the signal level estimator is fast when the value of the “attack time” and “release time” is large.
The signal quality of a hearing aid with respect to both speech intelligibility and listening comfort depends on both the speed of the signal level estimator and on the characteristics of the compressor itself.
The sound reproduced by the hearing aid will cause a pumping sensation when the change in gain has such a speed and magnitude, that the hearing aid wearer perceives a variation in sound level even in a steady sound environment. Typically the hearing aid wearer will in this case describe the reproduced sound as unsteady.
A compressor system with a slow signal level estimator normally results in good signal quality. However the signal level at the onset of e.g. a speech segment may become unacceptably loud because the sudden increase in sound input level is not immediately tracked by the compressor system because of the latency of the slow signal level estimator. Equally, the latency of the slow signal level estimator prevents appropriate amplification of a soft input signal following immediately after a sudden drop in sound input level (e.g. at the end of a spoken sentence). A fast signal level estimator will better trace the temporal characteristics of dynamic signals and hereby relieve the issues mentioned above for a slow signal level estimator. However, the signal quality generally decreases with a compressor based on a fast signal level estimator relative to a slow signal level estimator. Furthermore, the signal quality tends to degrade with increasing compression ratio, but on the other hand the compression ratio needs to be large enough to compress the dynamic range of the output signal adequately.
It is well known that the difference in speech intelligibility between normal hearing and hearing impaired subjects is larger in fluctuating noise than in stationary noise. In sound environments with highly fluctuating noise and a soft speaker it may therefore be advantageous to apply fast compression in order to straighten out the noise and hereby increase speech intelligibility for the hearing impaired.
A hearing aid with an improved compressor system providing greater flexibility with respect to the combination of the speed of the signal level estimators and the compression curve characteristics in order to improve the signal quality and speech intelligibility is thus desired.
2. Prior Art
European patent publication EP-A-1059016 describes a hearing aid device where the attack and release times are adjusted in response to the detected sound level to a relatively short duration providing fast gain adjustment at high input and/or output sound levels and to a relatively long duration providing slow gain adjustment at low input and/or output sound levels. By this method, the sound will be controlled with long attack and release times at low sound levels, at which the transfer function provides a compressor characteristic and the reproduced sound is very sensitive to pumping or vibrating sound effects when the gain varies with time. On the other hand, at elevated sound levels at which the reproduced sound approaches the clipping or pain threshold, the sound is controlled with short attack and release times.
Furthermore it is known in the art to have a multi-channel hearing aid with two separate compression systems working in parallel, where one system acts relatively slowly and has 15 channels, and the other system acts relatively faster and has 4 channels. The relative impact of the two compression systems is constantly adjusted. At soft to moderate sound levels the system responds more slowly, and with increasing sound levels the impact of the faster acting compression path increases.
The hearing aids described above do not allow the compressor system to be controlled by a slow signal level estimator at relatively high sound input levels (e.g. cocktail party situation), even though such a feature would be advantageous with respect to speech intelligibility.
WO-A1-03/081947 provides a method for a dynamic determination of time constants to be used in a detection of the signal level of an input signal of unknown level in an electric circuit. The method comprises the following steps: feed the input signal through an auxiliary level detection means that is reacting faster to changes in the input sound signal level than the detection of the signal level as a whole, feed either the input signal or the output of the auxiliary level detection means through a guided level detection means, which is arranged with a guided time constant, and where the guided level detection means outputs an estimate of the level of the input signal, analyze the outputs of the auxiliary and the guided level detector means and determine the time constant of the guided level detection means based on this analysis.
US-A1-2006/0233408 describes a hearing aid wherein the compressor adapts the attack and release time constants in response to input signal fluctuations or variations. In one embodiment, increases in the input signal level above the average signal level lead to decreased attack and release time constants.
Thus, none of the systems described above disclose the possibility of independently setting the compression curve characteristics for two compressors, which are working together, based on a slow and a fast signal level estimation respectively.
Therefore none of the systems described above allow the free adjustment of fast acting compression characteristics for optimizing to a specific sound environment without changing the input-output function prescribed by the fitting rationale.